GSM VoIP Gateway

Item No:MemberSC-385
Update time:2012-01-25
Description: Compatible with Asteristik Enable users to make 2 calls simultaeously from SIP VoIP to GSM or from GSM to SIP VoIP devices GSM SIP VoIP conversion 50 sets of LAN to GSM mobile routes setting 50 set of GSM mobile to LAN routes setting Voice response for setting and response (Dial in from GSM mobile) Standard SIP (RFC2543, RFC 3261) Protocol, communicate with other gateway or PC All function can be set up on web Dual band: 900/1800MHz Tri band: 900/1800/1900MHz Quad band: 900/1800/850/1900MHz

Item No:MemberSC-375
Update time:2007-11-19
Description: FEATURES: 50 sets of LAN --> MOBILE routes setting; 50 sets of MOBILE --> LAN routes setting -Support one stage diaing ** When LAN phone & SC-375 both register SIP proxy Server or Asterisk or VoipBuster,users can dial any destination number from LAN phone directly ** SIP Proxy Server, Asterisk need to have the route of destination number; VoipBuster need to have credit -Support assigned mode & free mode-two stage dialing Voice response for setting and status (dial in from mobile) Standard SIP(RFC2543RFC3261) protocol, communicates with other gateway or PC All functions can be set on web TCP/IP：IP/TCP/UDP/RTP/RTCP/、CMP/ARP/RARP/SNTP、DHCP/DNS Client、IEEE802.1P/Q、ToS/DiffServ、NAT Traversal、STUN、uPnP、IP Assignment、Static IP、DHCP、PPPoE Codec：G.711 u-Law、G.711 a-Law、G.723.1 (5.3k)、G.723.1 (6.3k)、G.729A、G.729A/B Voice Quality、VAD、CNG、AEC、LEC、Packet loss Example of Application: 3 Simple Steps in operation: Step 1. Change the Network setting if you need (Network/network setting) Step 2. Register SIP proxy Server or Asterisk or VoipBuster if you need (sip setting/service domain) Step 3. Set Route (request)